tag:blogger.com,1999:blog-460321274846352896.post5714662599481526283..comments2019-03-19T22:57:26.514-07:00Comments on Real Time Communications Bits: How are retransmissions implemented in WebRTCGustavo Garcíanoreply@blogger.comBlogger6125tag:blogger.com,1999:blog-460321274846352896.post-58318608613112255922018-08-20T01:44:53.181-07:002018-08-20T01:44:53.181-07:00Thanks very much for this post! I&#39;m currently ...Thanks very much for this post! I&#39;m currently implementing retransmissons for aiortc, my Python implementation of WebRTC and it answers a number of questions I had.Jeremy Lainéhttps://www.blogger.com/profile/00083196896063858525noreply@blogger.comtag:blogger.com,1999:blog-460321274846352896.post-90202564803589656182018-05-04T14:37:17.687-07:002018-05-04T14:37:17.687-07:00Hi, i think that i saw you visited my weblog thus ...<br />Hi, i think that i saw you visited my weblog thus i came to “return the favor”.I&#39;m trying to find things to enhance my site!I suppose its ok to use some of your ideas!! <a href="https://loginmaker.org/facebook/" rel="nofollow">facebook sign in</a> Patricia Howellhttps://www.blogger.com/profile/15424195959901517953noreply@blogger.comtag:blogger.com,1999:blog-460321274846352896.post-88795877083258773352017-11-15T11:05:55.383-08:002017-11-15T11:05:55.383-08:00Thank you for the post.
A typical video stream ha...Thank you for the post.<br /><br />A typical video stream has about 100 packets per second. Packet age of 10,000 is about 100 seconds old and 1,000 nack queue length is equal to at least 10 seconds. Is there a use case where 100 or 10 seconds old packets are still useful? Why don&#39;t they use 1 second limit as in the receiver side?<br /><br />Why do they keep a list of keyframes and not just the most recent one?<br /><br />I think they also check for the first packet in a keyframe and not if we got all the frame parts:<br />bool is_keyframe =<br /> packet.is_first_packet_in_frame &amp;&amp; packet.frameType == kVideoFrameKey;<br /><br />Seems like the webrtc source has moved:<br />https://chromium.googlesource.com/external/webrtc/+/master/modules/video_coding/nack_module.ccpablohttps://www.blogger.com/profile/07056804302996675811noreply@blogger.comtag:blogger.com,1999:blog-460321274846352896.post-16009686474177741572017-06-04T11:28:21.770-07:002017-06-04T11:28:21.770-07:00Thanks for sharing your honest experience. When I ...Thanks for sharing your honest experience. When I first took a look at my head shots,<br />I wasn’t too thrilled with mine but you’ve given me a new perspective!<br /><br /><a href="https://www.facebook.com/wwwvirtualedge/" rel="nofollow">Virtual Edge</a><br />Firozul Islamhttps://www.blogger.com/profile/14536461831501605697noreply@blogger.comtag:blogger.com,1999:blog-460321274846352896.post-91319783855912552922017-04-30T08:01:29.290-07:002017-04-30T08:01:29.290-07:00It is here, but don&#39;t expect it to be easy to ...It is here, but don&#39;t expect it to be easy to reuse in your app, it probably has many dependencies with rest of webrtc codebase: https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/audio_coding/neteq/Gustavo Garcíahttps://www.blogger.com/profile/09231386351348682181noreply@blogger.comtag:blogger.com,1999:blog-460321274846352896.post-90621543583876126962017-04-07T15:12:00.969-07:002017-04-07T15:12:00.969-07:00great post man!
Have a question. I&#39;m searching...great post man!<br />Have a question. I&#39;m searching the google code for the audio jitter buffer and not the video jitter buffer .<br />I want to make some tests and wrap this thing in my app. <br />do you know where i can start from?Unknownhttps://www.blogger.com/profile/01062272062536973240noreply@blogger.com